1. Field of the Invention
The present invention relates to information communications systems, and more particularly to a system and method for transmitting calls between networks which adhere to different communications protocols.
2. Background of the Related Art
A gateway serves as an interface between networks which use different protocols such as a public switched telephone network (PSTN), an Internet protocol (IP) network, or an integrated services digital network (ISDN). One type of protocol known as a session initiation protocol-telephone (SIP-T) protocol uses a session initiation protocol (SIP) to interface between a PSTN and an IP network. In one common application, the SIP-T is used as a protocol between media gateway controllers (MGC) when an ISDN user part (ISUP) message of the PSTN is transmitted to an asynchronous transfer mode (ATM)-based IP network.
FIG. 1 shows an open network structure for providing a voice-based ATM service. As shown, when a call set-up request is received from an originating PSTN network 10a, an origination MGC 30a transmits an SIP message to a terminating MGC 30b using the SIP-T protocol to request a call set-up from a terminating PSTN network 10b. The originating and terminating MGCs 30a and 30b control connection and release of an ATM bearer channel (64 Kbps) between the originating and terminating media gateways (MG) 40a and 40b. 
FIG. 2 shows a structure of the originating and terminating MGCs 30a and 30b. Each of these MGCs includes an ISUP block 31, a number translation (NTR) block 32, an SIP-T block 33, and a MEGACO block 34. The ISUP block 31 transmits and receives an ISUP message (IAM) required for a call set-up to and from the PSTN network (10a or 10b) through the SG 20a or 20b, and transmits a called party number contained in the corresponding ISUP message to the NTR block 32 to request a number translation. In addition, if a request for an additional digit required for the number translation is received from the NTR block 32, the ISUP block 31 collects the additional digit from the PSTN network (10a or 10b) and sends the number translation to the NTR block 32.
The NTR block 32 performs a number translation on the called party number according to the number translation request from the ISUP block 31 and provides an IP address of the terminating MGC 30b as a number translation result.
The SIP-T block 33 performs a Transmission Control Protocol (TCP)/User Data Protocol (UDP) connection to the terminating MGC 30b using the IP address of the terminating MGC 30b, which is obtained according to the number translation performed by the NTR block, and then performs connection and release functions for the bearer channel between the originating and terminating MGs 40a and 40b through the MEGACO block 34.
The MEGACO block 34 serves an interface between the SIP-T block 33 and the MG 40a or 40b. In performing this function, the MEGACO block transmits a command for controlling the connection and release of the ATM bearer channel to the MG, receives a response signal from the MG, and then provides the response signal to the SIP-T block 33.
FIG. 3 shows steps included in a call set-up process according to a related art which uses SIP-T overlap signaling in a voice-based open network. In this figure, the suffix ‘a’ is attached to reference numerals of the call originating side and the suffix ‘b’ is attached to reference numerals of the call terminating side. According to this process, when a caller connected to the PSTN network 10a dials a called party number (e.g., 450-7348), a switching system (not shown) of the PSTN network 10a inserts information into an ISUP message which includes an initial address message (IAM) and subsequent address messages (SAMs). The IAM message includes digits (‘450’) of the dialed number and the SAMs messages respectively include digit pairs (‘73’) and (‘48’) of the dialed number. These messages are then transmitted through the SG 20a to the originating MGC 30a. 
The ISUP block 31a of the originating MGC transmits the called digits (‘450’) contained in the IAM to the NTR block 32a, and transmits the number translation result provided from the NTR block 32a (i.e., an IP address of the terminating MGC 30B) together with the IAM to the SIP-T block 33a. 
The SIP-T block 33a is connected to the terminating MGC 30b via the TCP/UDP protocol using the IP address of the terminating MGC 30b, configures a new IAM (‘4507348’) by combining the IAM (‘450’) and the SAMs (‘73’ and ‘48’), and generates an SIP message (i.e., an INVITE message) containing the new IAM and transmits it to the terminating MGC 30b. In this respect, the SIP message includes the INVITE message and various status messages.
FIG. 4 shows a transfer structure of the INVITE message, which is one example of the SIP message. The INVITE message includes: a header containing a transmission medium request (TMR) for requesting a request line, a calling party number, a called party number, and a voice band width; a session description protocol (SCP) defining an attribute of the bearer channel; and the IAM.
After the SIP-T block 33a generates the INVITE message, it waits for receiving every digit before transmitting the INVITE message to the terminating MGC 30b. More specifically, the SIP-T block 33a collects the digits of the called party number until an inter-digit timer is terminated, and then includes the collected digits into the IAM and transmits the IAM containing the destination digits to the terminating MGC 30b. 
Returning to FIG. 3, when transmitting the INVITE message, in order to manage information on a call leg (Cleg), the SIP-T block 33a assigns a call leg number which is available for use (e.g., ‘1’), to the INVITE message.
Upon receiving the INVITE message, the SIP-T block 33b of the terminating MGC 30b transmits the called party number (‘4507348’) contained in the INVITE message to the NTR block 32b to request a number translation, so that the originating and terminating PSTN networks 10a and 10b can establish an ATM bearer channel through the MGs 40a and 40b. In addition, in order to prevent re-transmission of the INVITE message managed with the call leg (CLeg) ‘1’, the SIP-T block 33b transmits a status message (‘100’) indicating that the call is currently attempting a call setup to the SIP-T block 33a. 
After the NTR block 32b successfully performs the number translation, the SIP-T block 33b controls the terminating MG 40b through the MEGACO 34b with reference to the bearer attribute based on the SDP information of the INVITE message, so as to establish the ATM bearer channel between the originating and terminating MGs 40a and 40b. At the same time, SIP-T block 33b transmits the IAM message containing the called party number (‘4507348’) through the ISUP block 31b to the terminating PSTN network.
At this time, the switching system (not shown) of the terminating PSTN network performs a number translation on the received IAM message and transmits a call signal to a callee of the corresponding call. At the same time, the switching system transmits an address complete message (ACM), indicating that the called party number has been completely received and the call signal has been transmitted, through the ISUP block 31b to the SIP-T block 33b. 
Upon receiving the ACM, the SIP-T block 31b transmits a status message (‘180’) indicating that the call signal (Ring) is being transmitted for the CLeg ‘1’, to the SIP-T block 33a of the originating MGC 30a. And then, the SIP-T block 33a transmits the ACM message through the ISUP block 31a to the PSTN network 10a, so that the ring back tone is transmitted to the call sender by the PSTN network 10a. 
Thereafter, when the callee answers the call request, the terminating PSTN network 10b transmits an answer message (ANM) indicating that the call receiver has answered, through the ISUP block 31b to the SIP-T block 33b. The SIP-T block 31b then transmits a status message (‘200’) indicating that the callee has answered to the SIP-T block 33a of the MGC 30a. 
In response, the SIP-T block 33a transmits an acknowledgment ACK message to the SIP-T block 33b. At the same time, an ANM message is transmitted through the ISUP block 31a to the PSTN network 10a, so that a speech path is connected between the PSTN networks 10a and 10b through the ATM network 50. The caller and callee can then make a phone call.
In the call set-up method of the related art previously described, the originating MGC waits for receiving every digit before transmitting the INVITE message to the terminating PSTN network. That is, the originating MGC collects the digits of the called party number the caller has dialed until the inter-digit timer has expired and only then transmits the ISUP IAM message to the terminating PSTN network. This approach when applied using the SIP-T protocol has the following problem. Because the digits are not transmitted until after the inter-digit timer has expired are transmitted, and because the procedure for establishing the bearer channel therefore does not start until the inter-digit timer has expired, there is a disadvantage for the caller who can only hear a ring back tone after a delay caused by the time required for the inter-digit timer set-up process. A need therefore exists for an improved system and method for establishing a call, and more specifically one which does not introduce the delays contained in the related art.